[..] |
ASTERISK-27000: Asterisk 14.4.0 Make failes liblber-2.4.so.2: error adding symbols: DSO missing from command line |
ASTERISK-27001: res_pjsip: TLS connection not stable |
ASTERISK-27002: Error while trying to call back |
ASTERISK-27003: chan_sip doesn't send CONNECTEDLINE info over sip trunk |
ASTERISK-27004: Asterisk Playback with no answer on dahdi channel? |
ASTERISK-27005: The ARA (Real Time) work with ODBC connection + TCP - crash |
ASTERISK-27006: app_queue: Crash when hanging up with realtime queues |
ASTERISK-27007: Voice Issue |
ASTERISK-27008: res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space |
ASTERISK-27009: AMI Ping Action returns Permission denied |
ASTERISK-27010: Asterisk Crash with core dump |
ASTERISK-27011: ExtraChannel closed on AMI Redirect |
ASTERISK-27012: app_confbridge: ConfBridge sometimes does not play user name recording while leaving |
ASTERISK-27013: res_rtp_asterisk: Media can be hijacked even with strict RTP enabled |
ASTERISK-27014: configurable busy_timeout in sqlite backends |
ASTERISK-27015: RFC 3581 as client |
ASTERISK-27016: Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times. |
ASTERISK-27017: app_queue: Linear queue retries as many agents as were skipped |
ASTERISK-27018: Crash in res_rtp_asterisk.c |
ASTERISK-27019: CallerID not detected in India, two cases |
ASTERISK-27020: ARI breaks and doesn't send websocket messages when any party hangsup a call that is in a bridge |
ASTERISK-27021: GET /recordings/stored returns 500 Internal Server Error |
ASTERISK-27022: res_rtp_asterisk: Incorrect SSRC change for RTCP component |
ASTERISK-27023: res_rtp_asterisk: Deadlock when TURN session in use |
ASTERISK-27024: nat/external_media settings ignored in 14.4.1 |
ASTERISK-27025: channel / meetme: Fix missing parentheses |
ASTERISK-27026: res_ari: Crash when no ari.conf configuration file exists |
ASTERISK-27027: chan_sip: Channel Stuck when load reach to 500+ |
ASTERISK-27028: codec_opus: Build for ARM (pi, beagle bone, etc) |
ASTERISK-27029: ConfBridge Channel Locking |
ASTERISK-27030: Asterisk crashes with Segfault in res_ari |
ASTERISK-27031: res_pjsip: Unable to configure TLSv1.2 on TLS transport |
ASTERISK-27032: res_pjsip: TLS options do not handle empty values |
ASTERISK-27033: chan_dahdi - dialtone_detect not working |
ASTERISK-27036: res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@' |
ASTERISK-27037: Segfault in pjsip |
ASTERISK-27038: No audio while trying to connect two 3G sip phones via Asterisk |
ASTERISK-27039: chan_pjsip: Device state is idle when channel from endpoint is in early media |
ASTERISK-27040: MOH/Realtime: 'moh show classes' does not show classes defined in database |
ASTERISK-27041: Core/PBX: [patch] Deadlock between dialplan execution and application unregistration |
ASTERISK-27042: Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file |
ASTERISK-27043: Core/BuildSystem: Add defines to fix build with LibreSSL |
ASTERISK-27044: Documentation: Indicate that dialtone_detect and waitfordialtone are influenced by progzone option. |
ASTERISK-27045: Issue with Parsing Contact Header without Brackets and with additional HeaderParameters seperated with semicolon |
ASTERISK-27046: res_pjsip_transport_websocket: segfault in get_write_timeout |
ASTERISK-27047: res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be. |
ASTERISK-27048: Asterik segfault if ari.conf is not found |
ASTERISK-27049: asterisk-users mail list: User subscriptions getting auto-disabled due to excessive bouncing |
ASTERISK-27050: Crash on Transcoded Audio in PERIODIC_HOOK Function |
ASTERISK-27051: res_pjsip_mwi: unsolicited MWI has to be unsubscribed on deleting the endpoint's last contact |
ASTERISK-27052: Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network |
ASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer |
ASTERISK-27054: Crash In Queueing Frame |
ASTERISK-27055: Core show rtp payloads |
ASTERISK-27056: codec_opus responds with incorrect RTP Payload Type |
ASTERISK-27057: Seg Fault in ast_sorcery_object_get_id at sorcery.c |
ASTERISK-27058: Deadlock in ICE / SRTP |
ASTERISK-27059: res_stasis: Stolen channel references are leaking |
ASTERISK-27060: Comment typo format_g729.c |
ASTERISK-27061: res_pjsip: Crash/segfault during T.38 reinvite / negotiation |
ASTERISK-27062: documentation: Wiki for AMI actions is not user friendly |
ASTERISK-27063: Add support for systemd socket activation |
ASTERISK-27064: Wrapup doesn't work time-to-time. |
ASTERISK-27065: call hangup after leaving app_queue |
ASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode |
ASTERISK-27067: res_ari_channels: channel_state_invalid always leaks snapshot reference. |
ASTERISK-27068: app_voicemail: Add global option "imap_poll_logout" to specify post-polling disconnect |
ASTERISK-27069: Realtime MOH to remote MSQL instance not working |
ASTERISK-27070: res_pjsip: No SIP Re-INVITE on existing calls following reregistration on a different port |
ASTERISK-27071: chan_sip: MOH keeps playing on attended transfer to alcatel |
ASTERISK-27072: manager: AMI response contains order-important headers, violating AMI v2 spec |
ASTERISK-27073: manager: AMI "queues" action outputs freeform text that doesn't follow the AMI spec |
ASTERISK-27074: core_local: local channel data not being properly unref'ed and unlocked |
ASTERISK-27075: bridge: stuck channel(s) after failed attended transfer |
ASTERISK-27076: chan_pjsip: Add support for multiple streams |
ASTERISK-27077: chan_pjsip: Add support for multiple streams |
ASTERISK-27078: .wav to gsm format change |
ASTERISK-27079: PJSIP puts invalid data in SDP when using external_media_address |
ASTERISK-27080: res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled |
ASTERISK-27081: Dead Lock when DTMF pressed during ringing with d dial option |
ASTERISK-27082: res_fax: Error building JSON, Invalid UTF-8 string due to invalid FAX CSI string format |
ASTERISK-27083: Remove unused lifecycle callback in ast_sip_session_sdp_handler |
ASTERISK-27084: Reduce verbosity while loading PBX extensions. |
ASTERISK-27085: [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip |
ASTERISK-27086: Download content for certified-13.13-current is not updated for Certified Asterisk 13 -LTS |
ASTERISK-27087: res_ari: Crash in Asterisk 13.16 on Startup |
ASTERISK-27088: res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation |
ASTERISK-27089: pjsip: Switching from directmedia to T.38 doesn't use correct contact addresses. |
ASTERISK-27090: PJSIP: Deadlock using TCP transport |
ASTERISK-27091: pjsip: Crash when joining bridge |
ASTERISK-27092: [patch] app_queue: Add Priority to AMI QueueStatus |
ASTERISK-27093: ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting |
ASTERISK-27094: res_fax: Deadlock when using Local channels and fax gateway |
ASTERISK-27095: chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE |
ASTERISK-27096: res_rtp_asterisk: add a control frame for when dtls is established |
ASTERISK-27097: pjproject_bundled: We don't pass options needed for cross-compile to pjproject configure |
ASTERISK-27098: chan_pjsip: send indication for a video refresh for h264 |
ASTERISK-27099: Segfault in pjsip_message_ip_updater |
ASTERISK-27100: channel: ast_waitfordigit_full fails to clear flag in an error branch. |
ASTERISK-27101: res_pjsip: (IPv6 only, WSS) BYE is not emmited when Server Hangup Channel |
ASTERISK-27102: WSS PJSIP IPv6 Address type in SDP is not correct |
ASTERISK-27103: core: ast_safe_system command injection possible. |
ASTERISK-27104: After 10 minutes after reboot get Peer 'XXX' is now UNREACHABLE! |
ASTERISK-27105: [patch]core: when setting 'maxfiles' in asterisk.conf, a message is printed, even in rasterisk -x |
ASTERISK-27106: [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. |
ASTERISK-27107: how i can get input from user from ongoing conference using confbridge |
ASTERISK-27108: Crash using 'data get' CLI command |
ASTERISK-27109: No Outbound Audio After 10min |
ASTERISK-27110: RTP session is not fully destroyed on channel hangup |
ASTERISK-27111: Coredump with Asterisk 13.16.0 |
ASTERISK-27112: Core/BuildSystem: Cannot compile with AST_DEVMODE & OpenSSL >= 1.1 |
ASTERISK-27113: iLBC in Softmix yields choppy audio |
ASTERISK-27114: use background or playback play a wav file,and stop there,not continue |
ASTERISK-27115: Deadlock? db_sync_thread at db.c |
ASTERISK-27116: Voicemail ODBC seg fault issue |
ASTERISK-27117: core: Add support for timelen parsing to ast_parse_arg and ACO. |
ASTERISK-27118: res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE |
ASTERISK-27119: res_pjsip: parse/add msid attribute when webrtc is enabled |
ASTERISK-27120: Unsupported transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint |
ASTERISK-27121: res_pjsip_mwi: Memory leak on reload |
ASTERISK-27122: chan_iax2: On reload MWI taskprocessors keep adding up |
ASTERISK-27123: confbridge: Name recordings are left on filesystem |
ASTERISK-27124: app_playback.c:say_date_generic use timezonename parameter |
ASTERISK-27125: Core/Codec: Video playback extremely slow, around 1FPS |
ASTERISK-27126: "pjsip show aors" returns Memory Allocation Failure errors when using MariaDB ODBC connector |
ASTERISK-27127: configs: Erroneous load directive in sample configuration results in "Error loading module 'res_pjsip_multihomed.so'" |
ASTERISK-27128: [patch]res_stasis_snoop: When recording a snoop channel (using ARI) where no media is being received, no recording happens when theres no media |
ASTERISK-27129: ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait. |
ASTERISK-27130: Applications ARI: Unsubscribe action for deviceStates does not remove old subscriptions properly |
ASTERISK-27131: PJSIP: Registration problems with Polycom phones behind NAT |
ASTERISK-27132: Asterisk starts with multiple "undefined symbol" errors in pjsip modules |
ASTERISK-27133: res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use |
ASTERISK-27134: bridge_softmix: Reuse any removed streams for video |
ASTERISK-27135: CallerID not detected in India |
ASTERISK-27136: bridge_softmix: Don't reorder SFU streams |
ASTERISK-27137: chan_ooh323 per peer codec |
ASTERISK-27138: ooh323, no audio from Cisco CallManager Express ver.11.5 to asterisk |
ASTERISK-27139: pjsip: SDP attribute ptime ignored when Asterisk bridges two PJSIP channels with different ptime |
ASTERISK-27140: Memory leak in sorcery.c – pjsip subscriptions not expired |
ASTERISK-27141: Asterisk crashes when applying freePBX config |
ASTERISK-27142: sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5 |
ASTERISK-27143: bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. |
ASTERISK-27144: Fully Reproducible Dial Crash with Monitor i option + Playtones |
ASTERISK-27145: CISCO 7941G Phones with asterisk cannot receive Incoming call but can make an outgoing call to other extensions (Softphones or anyother brand like yealink) |
ASTERISK-27146: Crash during attended transfer |
ASTERISK-27147: Either asterisk or pjproject isn't re-using tcp connections (again) |
ASTERISK-27148: res_pjsip: Transports recreated wrongly, tls connections drop when config changes |
ASTERISK-27149: Segfault on reload |
ASTERISK-27150: Periodic crash in ast_channel_snapshot_create |
ASTERISK-27151: chan_sip.c/add_sdp() systematically adding m=video line |
ASTERISK-27152: Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash |
ASTERISK-27153: AMI Newexten event returns wrong event parameters |
ASTERISK-27154: Asterisk crash on stasis publish of channel leaving bridge event |
ASTERISK-27155: core dump might be pjsip related? |
ASTERISK-27156: Asterisk won't compile on Fedora 26 with devmode enabled. |
ASTERISK-27157: bridge: Crash when mapping streams |
ASTERISK-27158: [patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used |
ASTERISK-27159: Sounds: A few language sets are out of sync with the others after the recent update |
ASTERISK-27160: Hints generated by autohints=yes include Local/ channels and peers on remote end of trunks |
ASTERISK-27161: Local channel incorrectly changes its designated CallerIDNum |
ASTERISK-27162: [patch]chan_sip: Access incoming SIP REFER headers in the dialplan |
ASTERISK-27163: chan_sip: Dialplan function SIP_HEADERS() to complement SIP_HEADER(). |
ASTERISK-27164: [patch] Add IPv6 Support for DUNDi |
ASTERISK-27165: CDR: CDR(start,u) function won't work in cdr_custom config |
ASTERISK-27166: app_queue: Crash when handling hangup with incomplete data |
ASTERISK-27167: No data in custom cdr field in database when it set via Gosub |
ASTERISK-27168: alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table |
ASTERISK-27169: Google OAuth 2.0 support for XMPP / Motif |
ASTERISK-27170: pjproject: Unsafe usage of gethostbyname causing memory corruption |
ASTERISK-27171: Asterisk 15.0.0-Beta1 does not compile |
ASTERISK-27172: test_message: some tests fail |
ASTERISK-27173: Support for GMIME 3.0 |
ASTERISK-27174: res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical |
ASTERISK-27175: iax.conf demo peer is invalid |
ASTERISK-27176: test_abstract_jb: frames leak |
ASTERISK-27177: ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c |
ASTERISK-27178: Stream audio data monitor in Asterisk abstraction layer |
ASTERISK-27179: res_pjsip_session: Handling of 'msid' is incorrect |
ASTERISK-27180: channel: requester leaks joint_cap on success. |
ASTERISK-27181: GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting' |
ASTERISK-27182: bridge: Crash when mapping streams |
ASTERISK-27183: No data in custom cdr field in database when it overwrite via Gosub |
ASTERISK-27184: stream: Allow streams on a topology to be put into groups |
ASTERISK-27185: s3 bucket writable - asteriskconfig |
ASTERISK-27187: Asterisk keeps Crashing randomley |
ASTERISK-27188: PJSIP realtime |
ASTERISK-27189: Make --with-pjproject-bundled the default for Asterisk 15 |
ASTERISK-27190: Long delay when joining conference bridgw |
ASTERISK-27191: CDR: An extra character in the CDR "userfield" when it is populated with data from Sub called in Dial application |
ASTERISK-27192: res_pjsip: Loss of SIP registrations causing unavailable endpoints |
ASTERISK-27193: IPv6 receive address in message doesn't include brackets |
ASTERISK-27194: jitterbuffer: Does not handle case where translator returns null frame. |
ASTERISK-27195: chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets |
ASTERISK-27196: Asterisk certified/11.6-cert13 not releasing RTP ports |
ASTERISK-27197: Exceptionally long voice queue length queuing to Local/114@queue-dial-00001d8a;1 |
ASTERISK-27198: res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to yes |
ASTERISK-27199: Voicemail emails ignoring "serveremail" setting |
ASTERISK-27200: manager: hook event is not being raised |
ASTERISK-27201: Asterisk crashing when using SRTP |
ASTERISK-27202: If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed |
ASTERISK-27203: res_pjsip_pubsub: Crash when sending request due to subscription timeout |
ASTERISK-27204: [patch] app_queue: Wrong queue stat calculation |
ASTERISK-27205: res_rtp_asterisk: Asterisk crash with netlink error |
ASTERISK-27206: res_pjsip: No mechanism exists to limit endpoint identification to IP only |
ASTERISK-27207: XMPP OAuth not working due to inverted logic |
ASTERISK-27208: res_ari_bridges: Announcer channel in ChannelLeftBridge event |
ASTERISK-27209: Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used |
ASTERISK-27210: Getting segfault in res_pjsip.so and libasteriskpj.so.2 |
ASTERISK-27211: Bug in logger when transferring a call from a queue using ATTENDEDTRANSFER |
ASTERISK-27212: bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU |
ASTERISK-27213: asterisk crash randomly |
ASTERISK-27214: Recurring crash with memory corruption |
ASTERISK-27215: [patch]AMI : Add CancelAtxfer Action |
ASTERISK-27216: app_queue: does its check-makeannouncement-logic twice each head-caller-loop |
ASTERISK-27217: chan_sip: Asterisk crashing when subscription doesn't get set |
ASTERISK-27218: res_pjsip_sdp_rtp: Can't send packets to IPv4 address on IPv6 socket on some OSes |
ASTERISK-27219: asterisk crashing |
ASTERISK-27220: Enable CHANNEL function to get from and to tag from SIP Headers |
ASTERISK-27221: [patch] pbx_dundi: secretpath config setting is ignored |
ASTERISK-27222: core: Don't queue up multiple video update frames. |
ASTERISK-27223: chan_pjsip: unable to agree on audio codec with AVM Fritz!Box trunk (async rtp issue) |
ASTERISK-27224: Crash when freeing log message |
ASTERISK-27225: Crash when freeing dtls_cfg->cafile |
ASTERISK-27226: core: Audit frame types that shouldn't be queued multiple times |
ASTERISK-27227: Frequent Crash (Possibly due to unbundled PJSIP) |
ASTERISK-27228: bridging: Two party bridge where the first party was dialing a third party causes a variety of weird behavior |
ASTERISK-27229: bridge: Old channel video source not set to NULL after unref |
ASTERISK-27230: PJSIP Destroyed timer being called causing segfault |
ASTERISK-27231: res_rtp_asterisk: Allow remote SSRC to change due to renegotiation |
ASTERISK-27232: When in queue on g722 with interruptions, music on hold can get stuck and no longer play |
ASTERISK-27233: enrredo |
ASTERISK-27234: Crash on hangup |
ASTERISK-27235: Crash when freeing frame in bridge |
ASTERISK-27236: Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive |
ASTERISK-27237: when emailing a voicemail we need the DID of the trunk call came in on |
ASTERISK-27238: Bridging: Crash freeing a frame that's already been freed |
ASTERISK-27239: PJSIP losing RTP connection (no more audio) |
ASTERISK-27240: SHELL function it's returning wrong value |
ASTERISK-27241: libc segfault upon entry into app_directory |
ASTERISK-27242: Asterisk stops responding to packets |
ASTERISK-27243: contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax |
ASTERISK-27244: Realtime mapping for 'ps_aors' found to engine 'xxxx', but the engine is not available |
ASTERISK-27245: Access to AMI always raise a tls exception even if tls is disable |
ASTERISK-27246: app_directory.c doesn't create /var/spool/asterisk/voicemail subdirectories after server move |
ASTERISK-27247: Asterisk not responding for 5 to 15 seconds |
ASTERISK-27248: [patch]external_media_address and external_signaling_address don't always honor localnet |
ASTERISK-27249: How to check running calls on Inbound Route |
ASTERISK-27250: chan_pjsip: asymmetric_rtp_codec=no does not seem to be working anymore with Asterisk 13.17.1 |
ASTERISK-27251: chan_sip doesn't honour rtptimeout setting |
ASTERISK-27252: RTP: One way audio with direct media and strictrtp=yes. |
ASTERISK-27253: [patch] libsrtp-2.1.x support |
ASTERISK-27254: alembic: prune_on_boot fix erroneous |
ASTERISK-27255: alembic: Add support for Microsoft SQL server |
ASTERISK-27256: Segfault while sending unsolicited MWI |
ASTERISK-27257: bridge_native_rtp: half-way direct media when using early bridging |
ASTERISK-27258: PJSIP issues and possible memory leaks???? |
ASTERISK-27259: chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller |
ASTERISK-27260: [pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36 |
ASTERISK-27261: Memory Leak when using ARI in json.c and stasis_channels.c |
ASTERISK-27262: res_ari: Leaking eventfds when using ARI Dial |
ASTERISK-27263: Attended transfer by SNOM not working : REFER rejected by Asterisk |
ASTERISK-27264: res_pjsip_session: Crashes after sending PRACK and receiving 200 OK |
ASTERISK-27265: with PJSIP alert info is not working to distinguish internal, external and group calls |
ASTERISK-27266: cel and channel(userfield) |
ASTERISK-27267: Asterisk crashes after module reload |
ASTERISK-27268: pjsip: Lockup PJSIP Invite+Reg Processing Taskprocessor w/ distributor backup |
ASTERISK-27269: asterisk doesn't work with latest rhel/centos 7.4 kernel |
ASTERISK-27270: cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured |
ASTERISK-27271: astobj2.c:131 INTERNAL_OBJ: FRACK!, Failed assertion bad magic number 0x0 for object 0x166bdf8 (0) |
ASTERISK-27272: res_config_pjsql, contrib: Invalid input value for enum yesno_values: "true" |
ASTERISK-27273: app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command |
ASTERISK-27274: RTCP needs better packet validation to resist port scans. |
ASTERISK-27275: PJSip uses wrong codec |
ASTERISK-27276: pjsip crash given aor with no contacts |
ASTERISK-27277: bridge: Renegotiate if source stream changes. |
ASTERISK-27278: [patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE |
ASTERISK-27279: Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability |
ASTERISK-27280: app_dial: Q Option Unexpectedly Removing Reason:SIP cause Header |
ASTERISK-27281: Increased memory usage when using Sorcery Caching |
ASTERISK-27282: Confbridge leave announcement fails due to missing file |
ASTERISK-27283: Realtime config fail with PostgreSQL version before 9.1 |
ASTERISK-27284: Status of RFC 3323 and PJSIP |
ASTERISK-27285: AstDB Locks taking a while to unlock |
ASTERISK-27286: Add the ability to read the media file type from HTTP header for playback |
ASTERISK-27287: Cyclic reference between res_pjsip and res_pjsip_session |
ASTERISK-27288: res_fax:Fax gateway framehook not moved properly upon faxdetect masquerade |
ASTERISK-27289: A codeblock that maintains a bug,but maybe the codeblock will never run |
ASTERISK-27290: res_pjsip: PIDF contact field has malformed/invalid XML |
ASTERISK-27291: I CAN'T MAKE CALL IN ASTERISK |
ASTERISK-27292: Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes) |
ASTERISK-27293: Queue Application - gosub and maco parameter functionality seem broken |
ASTERISK-27294: res_pjsip: Asterisk has to be restarted to apply new external settings |
ASTERISK-27295: Contact is improperly translated after d178f497 |
ASTERISK-27296: [patch] False positive busy checks when icalendar's recurrence-id mechanism is involved |
ASTERISK-27297: ChanSpy attaching to wrong Channel with similar Name |
ASTERISK-27298: Problem with expires on pjsip / outbound-publish |
ASTERISK-27299: Asterisk Hangs with Bad file descriptor on read() |
ASTERISK-27300: Asterisk crashes randomly (FRACK!, chan_sip) |
ASTERISK-27301: [patch] app_queue: Music On Hold for real-time queues is not reset to default |
ASTERISK-27302: Segfault grp_lock_acquire while processing endpoint event |
ASTERISK-27303: [patch] chan_sip: Choppy MoH when channel on hold and bridge is native_rtp |
ASTERISK-27304: Registration with digest authentication in PJSIP fails if a username contains symbol @ |
ASTERISK-27305: res_ari: Memory leaks in ARI when using Content-Type: application/json |
ASTERISK-27306: chan_pjsip: Cannot be tested for memory leaks. |
ASTERISK-27307: AMI Action 'Command' only returns the last line |
ASTERISK-27308: res_pjsip_t38: Faxing does not work under FreeBSD |
ASTERISK-27309: Feature Parity with chan_sip |
ASTERISK-27310: Advice Of Charge (AOC) Support |
ASTERISK-27311: Call Completion Supplementary Services (CCSS) Support |
ASTERISK-27312: Outbound SUBSCRIBE Support |
ASTERISK-27313: Warning message netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null) |
ASTERISK-27314: chan_sip: Crash/Deadlock with realtime peers (MySQL) |
ASTERISK-27315: Calendar Feature On Asterisk |
ASTERISK-27316: stream: Leak of format_cap |
ASTERISK-27317: vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. |
ASTERISK-27318: res_pjsip_mwi: uninitialized value from ast_strings_match |
ASTERISK-27319: (Security) Function in PJSIP 2.7 miscalculates the length of an unsigned long variable in 64bit machines |
ASTERISK-27320: Testing Remote Clone (Disregard - will be deleted) |
ASTERISK-27321: Asterisk Crashing with FRACK Errors and Serious Network Trouble |
ASTERISK-27322: [New Feature] Add mute and DTMF passthrough to ARI add channel to bridge |
ASTERISK-27323: asterisk ip-pbx |
ASTERISK-27324: [patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS |
ASTERISK-27325: app_queue: Agent stays in use after call is parked using DTMF |
ASTERISK-27326: Asterisk crashes from time to time |
ASTERISK-27327: glibc detected |
ASTERISK-27328: Missing openssl dependencies in res_rtp_asterisk and tcptls |
ASTERISK-27329: PJSIP T.38 UDPL transmission error |
ASTERISK-27330: Asterisk crashes sometimes on destruction of RTCP message payload |
ASTERISK-27331: Jenkins: Reimplement REF_DEBUG testing. |
ASTERISK-27332: Asterisk fails to configure on MacOS Sierra |
ASTERISK-27333: sip_to_pjsip not correctly handling disallow=all directive |
ASTERISK-27334: Pjsip Performance issues with qualify enabled |
ASTERISK-27335: CDR performance needs improvement. |
ASTERISK-27336: Support for 44.1khz/48khz WAV files |
ASTERISK-27337: chan_sip: Security vulnerability with client code header (revisited) |
ASTERISK-27338: Excessive refcount 100000 reached on ao2 object |
ASTERISK-27339: [patch] Crash on ast_ssl_teardown when stopping. |
ASTERISK-27340: backtrace.c: Crash due to double-free. |
ASTERISK-27341: [patch] res_pjsip_session: SIP/SDP origin (o=) contains local address. |
ASTERISK-27342: [patch] libpri: Calling Party Subaddress extended octet 3 encoding incorrectly handled |
ASTERISK-27343: Fails to build in FreeBSD due to sys/sysmacros.h not existing there |
ASTERISK-27345: res_pjsip_session: RTP instances leak on 488 responses. |
ASTERISK-27346: res_xmpp: Crash if OAuth 2.0 is used before curl is loaded |
ASTERISK-27347: [patch] pjproject_bundled: Disable TCP/TLS keep-alives. |
ASTERISK-27348: [patch]contrib/scripts: add a way to migrate from chan_sip to chan_pjsip realtime |
ASTERISK-27349: Failed to join Bridge when calling ConfBridge from applicationmap |
ASTERISK-27350: app_macro deprecation |
ASTERISK-27351: iLBC Codec Mismatch |
ASTERISK-27352: Asterisk ignoring top codec in 200 OK |
ASTERISK-27353: H323 audio starts with a delay of 2 seconds. |
ASTERISK-27354: bridge_softmix: When a channel leaves add in any missing participant streams |
ASTERISK-27355: Upgrade bundled PJPROJECT to 2.7 |
ASTERISK-27356: [patch] libsrtp-2.x.x + AES-GCM support |
ASTERISK-27357: PJSip endpoints lose AoRs when enabling ODBC |
ASTERISK-27358: res_pjsip: endpt_send_request leaks |
ASTERISK-27359: pjproject bundled: Don't disable assertions when --enable-dev-mode is used. |
ASTERISK-27360: Audio file from ffmpeg pops in AGI (but not dialplan) |
ASTERISK-27361: Attended transfer crashes in Asterisk 13.17.2 |
ASTERISK-27362: Wrong BLF for "ringing" in Snom/Microsip |
ASTERISK-27363: res_http_websocket: Wrong LocalAddress (it is equal to RemoteAddress) |
ASTERISK-27364: channel: Crash when fax gateway is in use with PJSIP |
ASTERISK-27365: [patch] chan_sip: Crypto attribute not last but first on SDP media level. |
ASTERISK-27366: Asterisk Turkish Language Set Problem |
ASTERISK-27367: logger: We should call rotate_file on huge logs even if rotatestrategy is none. |
ASTERISK-27368: Can't send 10 type frames with PJSIP |
ASTERISK-27369: Bridge() dialplan application fails without setting BRIDGERESULT channel variable |
ASTERISK-27370: [Patch] Bridge() dialplan application fails without setting BRIDGERESULT channel variable |
ASTERISK-27371: Serious Network Trouble Errors with TCP Enabled |
ASTERISK-27372: ARI: Node ARI client broken in latest versions of 13 and 14 |
ASTERISK-27373: ringinuse=no is ignored in queue using asterisk realtime |
ASTERISK-27374: alembic: PJSIP scripts are missing column bundle in ps_endpoints table |
ASTERISK-27375: Jenkins: Add beanstalkd client libraries to slaves |
ASTERISK-27376: Crash Asterisk chan_sip rtcp_mux |
ASTERISK-27377: Typo in CHANNEL(dtmf_features) usage documentation |
ASTERISK-27378: Modules: Fix issues with CLI completion. |
ASTERISK-27379: stream: Allow streams on a topology to be put into groups |
ASTERISK-27380: ast_coredumper: allow pointing out the asterisk binary explicitly |
ASTERISK-27381: Crash inside opus codec |
ASTERISK-27382: crash after an invalid rtcp packet from GT48 FXS gateway |
ASTERISK-27383: astdb: Document performance issues and alternatives |
ASTERISK-27384: DNS related segfault using pjsip and multiple DNS SRV records |
ASTERISK-27385: channel.c: Exceptionally long queue length queuing |
ASTERISK-27386: asterisk seems to miss udp port change in ok message |
ASTERISK-27387: Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more |
ASTERISK-27388: Action QueueStatus - AMI, causing "Segmentation fault" |
ASTERISK-27389: Optional API modules should not allow unload. |
ASTERISK-27390: Audit menuselect module dependencies |
ASTERISK-27391: Regression: Deadlock between AOR named lock and pjproject grp lock |
ASTERISK-27392: configure --with-download-cache does not work with relative paths. |
ASTERISK-27393: res_pjsip: Crash occurs when an empty contact read from astdb or database |
ASTERISK-27394: [patch] tcptls: Print notice when TLS is enabled but not configured. |
ASTERISK-27395: srtp: Add support for ephemeral DTLS certificates |
ASTERISK-27396: Hangup not processed until voicemail timeout |
ASTERISK-27397: res_srtp / res_pjsip_sdp_rtp: New key on answer to reinvite not applied correctly |
ASTERISK-27398: No joint capabilities with video and audio-only streams |
ASTERISK-27399: ChanIsAvail returns PJSIP peer that is not working |
ASTERISK-27400: Migracion telefonia fija |
ASTERISK-27401: Random crashs during the automatic calendars refresh |
ASTERISK-27402: AMI generating incorrect ContactStatus events |
ASTERISK-27403: Username Mismatch on SIP Trunk |
ASTERISK-27404: DEBUG_FD_LEAKS does not record socketpair, timerfd_create or eventfd. |
ASTERISK-27405: Audio distortion after 3f7d0b63 |
ASTERISK-27406: Infinite loop when out of ports and rtpstart value is odd |
ASTERISK-27407: Boolean types in sorcery that map to yes/no Enum values are not correctly mapped to database |
ASTERISK-27408: Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp |
ASTERISK-27409: Asterisk got crashed : Program terminated with signal 6, Aborted. |
ASTERISK-27410: testsuite: channels/pjsip/acl_call backtrace |
ASTERISK-27411: pjsip: TCP connections may not be destroyed |
ASTERISK-27412: core: Audiohook freeing interpolated frame when it shouldn't. |
ASTERISK-27413: Add cache_media_frames debugging option. |
ASTERISK-27414: Inbound route to misc destination as destination doesn't record the name of the src to CDR |
ASTERISK-27415: asterisk.conf: Setting astctl without setting astrundir is ineffective. |
ASTERISK-27416: Can't load res_corosync.so module on Asterisk 13.18.2 |
ASTERISK-27417: Endpoints not getting registered |
ASTERISK-27418: app_confbridge: "core show profile bridge" does not output "sfu" when video_mode is sfu |
ASTERISK-27419: Support RFC 5009 P-Early-Media |
ASTERISK-27420: Asterisk crash - Unexpected error 9 on netlink descriptor 99 |
ASTERISK-27421: RTP source learning not working with devices that have some clock issues |
ASTERISK-27422: What protocol is used to send DTMF. |
ASTERISK-27423: app_record: We set the RECORD_STATUS channel variable before closing the file |
ASTERISK-27424: PJSIP T.38 Re-Invites fail between 2 extensions running Fax Voip FSP Windows Fax Service Provider |
ASTERISK-27425: Calls are not billed correctly by a2billing - Asterisk 12 and 13 |
ASTERISK-27426: chan_console: cannot read and write at the same time with alsa backend |
ASTERISK-27427: add fullchain.pem support for builtin http server |
ASTERISK-27428: On Asterisk 15.1.0, AlarmReceiver() Application not flushing detected event to the event-file |
ASTERISK-27429: res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should |
ASTERISK-27430: README refers to security documents that do not exist. |
ASTERISK-27431: Asterisk fails to build when openssl headers are not installed. |
ASTERISK-27432: configure ignores option --disable-asteriskssl |
ASTERISK-27433: Call Monitor doesn't work with native bridge |
ASTERISK-27434: [patch] chan_sip/ICE: Square brackets around IPv6 addresses. |
ASTERISK-27435: [patch] configure: pjsip_evsub_set_uas_timeout not found. |
ASTERISK-27436: rtp openssl errors |
ASTERISK-27437: [patch] ICE: server-reflexive candidates (srflx) with Dual-Stack. |
ASTERISK-27438: Custom variable at endpoint definition are duplicated |
ASTERISK-27439: PJSIP not correct work with expire time |
ASTERISK-27440: Strictrtp has issues to qualify video rtp streams |
ASTERISK-27441: PJSIP: Forked INVITE SDP negotiation gets one way audio. |
ASTERISK-27442: pjsip: 183 without To tag does not negotiate media |
ASTERISK-27443: Add a dynamic feature on callee channel |
ASTERISK-27444: CDR Error on ConfChannel |
ASTERISK-27445: ARI: Updating a bridge gives wrong error message. |
ASTERISK-27446: No 'h' extension run when call transferred by caller completes |
ASTERISK-27447: MOH: Crash scanning MOH files using realtime. |
ASTERISK-27448: [patch] Add ability to send progress inband by setting a channel variable instead of fixed config per endpoint |
ASTERISK-27449: [PATCH] When failing to acquire target during attended transfer, display wanted extension |
ASTERISK-27450: PJSIP Path header |
ASTERISK-27451: I get error while installing asterisk version 13 after executing make command. |
ASTERISK-27452: Security: chan_skinny: Memory exhaustion if flooded with unauthenticated requests |
ASTERISK-27453: RTP: Blind transfer direct media scenario results in one way audio. |
ASTERISK-27454: res_http_post: Don't require GMIME_MAJOR_VERSION |
ASTERISK-27455: PJSIP AOR Path header |
ASTERISK-27456: app_voicemail: Add new object for VoicemailUserEntry |
ASTERISK-27457: chan_sip: Guests disallowed via TCP (or TLS) if existing peer from same IP. |
ASTERISK-27458: Strict RTP protection Issue |
ASTERISK-27459: WebRTC |
ASTERISK-27460: CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... |
ASTERISK-27461: 3PCC patch for AMI "SIPnotify" |
ASTERISK-27462: Asterisk crashed |
ASTERISK-27463: Asterisk fails randomly |
ASTERISK-27464: how get calee ip address on pjsip channel |
ASTERISK-27465: CLI Completion Not Working |
ASTERISK-27466: Asterisk 14.7.3 Crash SIP - chan_sip.c: FRACK!, Failed assertion bad magic number 0x0 |
ASTERISK-27467: pjsip_options: qualify_frequency sometimes not applied on reload |
ASTERISK-27468: Asterisk crash on agi execute CELGenUserEvent with json data |
ASTERISK-27469: [patch] app_queue: While using queues with realtime, setting back to an empty announce don't stop the old announce to be played |
ASTERISK-27470: Add new object for VoicemailUserEntry |
ASTERISK-27471: res_pjsip_pubsub: Crash when accepting inbound subscription due to no memory pool |
ASTERISK-27472: 401 Unauthorized from INVITE not generating security event |
ASTERISK-27473: FXO incoming/outgoing issue |
ASTERISK-27474: PJSIP TLS Unstable |
ASTERISK-27475: codec_opus requires libcurl |
ASTERISK-27476: One-way audio when dual media streams are present in one RTP session |
ASTERISK-27477: Chan_pjsip does not support unauthenticated OPTIONS ping |
ASTERISK-27478: PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI. |
ASTERISK-27479: [patch] app_queue: While using queues with realtime, setting back to an empty announce don't stop the old announce to be played |
ASTERISK-27480: Security: Authenticated SUBSCRIBE without Contact crashes asterisk |
ASTERISK-27481: Asterisk crashes when receiving REFER message on PJSIP channel |
ASTERISK-27482: SRTCP unprotect failed because of authentication failure |
ASTERISK-27483: Allow wrapuptime to be set for each queue member |
ASTERISK-27484: Crash in attempting to bridge channels |
ASTERISK-27485: asterisk crashes when doing playback of a SLIN file and chanspy with eagi |
ASTERISK-27486: Static members lost "pause" on asterisk restart or module unload/load |
ASTERISK-27487: MESSAGE |
ASTERISK-27488: core: If frame with unnegotiated format is read crash will occur |
ASTERISK-27489: Realtime/ODBC voicemail concurrency regression |
ASTERISK-27490: chan_console: 'set active' fails to work |
ASTERISK-27491: res_pjsip_endpoint_identifier_ip only matches against header if match by ip fails |
ASTERISK-27492: Testsuite: Create a branching structure |
ASTERISK-27495: DNS: Unexpected rr_type can cause crash |
ASTERISK-27498: ICE candidate parser - ICE foundation parsing too short |
ASTERISK-27499: Make build of Asterisk reproducible, if so required |
ASTERISK-27514: Link error of astdb2bdb with Kernel 4.14.8 |
ASTERISK-27515: prune_on_boot field missing in ps_contact table |
ASTERISK-27529: Endpoint with TLS lost inbound registration |
ASTERISK-27530: not getting call rejecting event |
ASTERISK-27531: Compiler optimizations can break module load sequence. |
ASTERISK-27532: ERROR[5383][C-00003919] astobj2.c: Excessive refcount 100000 reached on ao2 object 0x142f648 |
ASTERISK-27533: res_pjsip: Large number of inbound/outbound registrations exhausts memory pool |
ASTERISK-27534: chan_sip: Assumes iostream is non-NULL when it may not be |
ASTERISK-27535: Asterisk crash with segmentation fault in libpthread-2.19.so |
ASTERISK-27536: res_pjsip.c: Error 70010 'Too many objects of the specified type (PJ_ETOOMANY)' sending OPTIONS request to endpoint |
ASTERISK-27537: res_pjsip: Add new AMI Action for PJSIPShowAors |
ASTERISK-27538: chan_sip: Stuck channels |
ASTERISK-27539: 'cdr submit' fails: batch mode not enabled. |
ASTERISK-27540: test_config: Missing XML documentation |
ASTERISK-27541: app_queue: Queue paused reason was (big number) secs ago when reason is set |
ASTERISK-27542: app_queue: When "queue show" CLI command is executed a crash occurs |
ASTERISK-27543: Segfault in pjsip_expires_hdr_create |
ASTERISK-27544: have 6 asterisk servers, working perfectly, suddenly all stop responding or delaying the responses for a few minutes causing havoc |
ASTERISK-27545: chan_sip: improper handling of Re-invites between IPv4 and IPv6 and vice-versa |
ASTERISK-27546: say: 'Q' and 'q' in English time formats only check the past |
ASTERISK-27547: res_pjsip: Add new AMI Action for PJSIPShowAuths |
ASTERISK-27548: res_pjsip_endpoint_identifier_ip only matches against "generic string" headers |
ASTERISK-27549: [patch] translate: Avoid absolute value on unsigned substraction. |
ASTERISK-27550: [patch] bridge_softmix: Avoid warning about an uninitialized variable. |
ASTERISK-27551: [patch] ooh323cDriver: Fix typo in header guard. |
ASTERISK-27552: [patch] chan_ooh323: Limit outgoinglimit to positive values as intended. |
ASTERISK-27553: [patch] res_curl: Avoid error message on unload. |
ASTERISK-27554: res_pjsip_rfc3326: Order of 'Reason' headers break many endpoints |
ASTERISK-27555: [patch] install_prereq: Update Debian/Ubuntu libraries. |
ASTERISK-27556: consumes high process in core show task processors [stasis-core & stasis-core-processors] |
ASTERISK-27557: [patch] clang 5.0: implicit conversion to char changes value to negative. |
ASTERISK-27558: [patch] codec_gsm: Avoid shifting a negative signed value. |
ASTERISK-27559: [patch] editline: Avoid comparison between pointer and zero character constant. |
ASTERISK-27560: [patch] clang 5 does not know -Wno-format-truncation |
ASTERISK-27561: Store CallerID name for the ENTERQUEUE queue log event |
ASTERISK-27562: ExtenSpy with whisper moves to wrong extension |
ASTERISK-27563: pjsip modules always get -O2 even when DONT_OPTIMIZE is set |
ASTERISK-27566: res_pjsip_session: Improve WebRTC interop with bundling during renegotiation |
ASTERISK-27567: app_queue.so segfault |
ASTERISK-27568: PJSIP: Crash during SIP attended transfer. |
ASTERISK-27569: Not answering call on Polarity reversal Event |
ASTERISK-27570: tests/channels/SIP/path: Sporadic failure due to DNS resolution |
ASTERISK-27571: res_pjsip: If SIP response is received during shutdown a crash may occur |
ASTERISK-27572: cdr_mysql creates empty records if reconnects when mysql was not up on module load |
ASTERISK-27573: voice mail sound deleted after |
ASTERISK-27574: Asterisk rejecting invites == 416 Unsupported URI scheme |
ASTERISK-27575: menuselect : remove obsolete TRACE_FRAMES compiler flag |
ASTERISK-27576: [patch] res_config_pgsql: Avoid typecasting an int to unsigned char. |
ASTERISK-27577: [patch] chan_ooh323: Avoid typecasting an int to unsigned short. |
ASTERISK-27578: [patch] app_osplookup.c: Avoid a format truncation. |
ASTERISK-27579: app_bridgewait: ring tone generation not stopped with entertainment disabled |
ASTERISK-27580: [patch] lpc10: Array may be uninitialized. |
ASTERISK-27581: Add new AMI Action for PJSIPShowContacts |
ASTERISK-27582: Segmentation fault occurs in Asterisk with an invalid SDP media format description |
ASTERISK-27583: Segmentation fault occurs in asterisk with an invalid SDP fmtp attribute |
ASTERISK-27584: Internal pjproject build doesn't disable bcg729 |
ASTERISK-27585: [patch] BuildSystem: Resolve resolv.h not via Generic but Particular Header-Check. |
ASTERISK-27586: ConfBridge AMI events missing documentation. |
ASTERISK-27587: Asterisk webrtc con JSSIP dont close connection |
ASTERISK-27588: Console Print Crash |
ASTERISK-27589: [patch] BuildSystem: Avoid $EUID and use id -u instead. |
ASTERISK-27590: res_corosync doesn't load in builds after version 13.17.1 |
ASTERISK-27591: Frack errors in stasis.c and memory leakage |
ASTERISK-27592: [patch] BuildSystem: Detect external library Lua in version 5.3. |
ASTERISK-27593: [patch] BuildSystem: In OpenBSD, xmlstarlet is xml. |
ASTERISK-27594: [patch] BuildSystem: Invoke install not in GNU but POSIX style. |
ASTERISK-27595: [patch] BuildSystem: Invoke ldconfig with previous paths. |
ASTERISK-27596: [patch] BuildSystem: Use the detected name for MD5 everywhere. |
ASTERISK-27597: AMI Queuestatus not working (with realtime queue) |
ASTERISK-27598: [patch] install_prereq: Support package manager DNF. |
ASTERISK-27599: [patch] install_prereq: Update RHEL/CentOS/Fedora libraries. |
ASTERISK-27600: [patch] BuildSystem: Allow make clean all again. |
ASTERISK-27601: chan_sip. unregister |
ASTERISK-27602: [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory. |
ASTERISK-27603: [patch] install_prereq: Download latest Jansson. |
ASTERISK-27604: chan_sip: turning on srtp results in one way audio in Asterisk 15 |
ASTERISK-27605: ICE: Not configurable by FQDN for externhost (chan_sip) / external_media_address (chan_pjsip). |
ASTERISK-27606: BuildSystem: declare -A assumes shell Bash. |
ASTERISK-27607: [patch] res_config_mysql: Avoid the header mysql_version.h. |
ASTERISK-27608: Asterisk threads consume cpu usage waiting on pj_ioqueue_poll |
ASTERISK-27609: sip_tcptls_read: SIP TCP/TLS server has shut down after 120s |
ASTERISK-27610: app_amd.so returning TOOLONG before reaching the timeout |
ASTERISK-27611: Giffy charts embedded in the Wiki now give an error |
ASTERISK-27612: Subscriptions Persist After Expiration and TCP/TLS Disconnect |
ASTERISK-27613: News & Security Advisories RSS Feeds |
ASTERISK-27614: res_pjsip_session: SDP origin does not use resolved address |
ASTERISK-27615: Dialplan deadlock when connection to external SQL server is lost |
ASTERISK-27616: chan_sip locks up during reload under Asterisk 13 / 15 (but not 11) |
ASTERISK-27617: Frack (crash), excessive refcount during Jitterbuffer operation |
ASTERISK-27618: Crash occurs when sending a repeated number of INVITE messages over TCP or TLS transport |
ASTERISK-27619: Build System: Require compiler to provide built-in support for atomic references. |
ASTERISK-27620: New module loader aborts startup if a required module declines load. |
ASTERISK-27621: (null) string tailing after AsyncAGIEnd AMI event |
ASTERISK-27622: empty voicemail.conf required for ARA (realtime) voicemail to leave message |
ASTERISK-27623: get_rdnis: Huh? Not an RDNIS SIP header (tel:xxx |
ASTERISK-27624: WebRTC regression with Asterisk 15 and Chrome 64 to receive calls |
ASTERISK-27625: channels: CHECK_BLOCKING is ineffective |
ASTERISK-27626: PJSIP asserts in "tsx_on_state_completed_uas" |
ASTERISK-27627: SDP payload sets ‘127.0.0.1’ instead of ‘media_address’-Parameter |
ASTERISK-27628: Asterisk stops processing events from the network |
ASTERISK-27629: [patch] headers: Replace typeof with __typeof__. |
ASTERISK-27630: [patch] editline: Avoid shifting a negative signed value. |
ASTERISK-27631: [patch] BuildSystem: Do not warn when bash is not installed. |
ASTERISK-27633: PJSIP crash phone using UDP on TCP endpoint in ast_sip_failover_request |
ASTERISK-27634: Determine if the internal editline and stdtime libraries are still relevant |
ASTERISK-27635: [patch] app_voicemail: Avoid always true warnings with clang. |
ASTERISK-27636: One way audio on Chrome 64 webrtc chan_sip |
ASTERISK-27637: [patch] BuildSystem: Enable autotools in FreeBSD. |
ASTERISK-27638: BuildSystem: Enable Lua in FreeBSD. |
ASTERISK-27639: [patch] BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD. |
ASTERISK-27640: SUBSCRIBE message with a large Accept value causes stack corruption |
ASTERISK-27641: BuildSystem: Enable Better Backtraces in FreeBSD. |
ASTERISK-27642: [patch] backtrace: Avoid -Wlogical-not-parentheses. |
ASTERISK-27643: SIP INVITEs to non-5060 port still use 5060 (Outbound Calls failing) |
ASTERISK-27644: Non stop show this Warning LOG " dsp.c:1421 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833"How can i solved it??? |
ASTERISK-27645: Asterisk 13.18 crash in pj_atomic_dec_and_get |
ASTERISK-27646: ICE fails with no candidate nominated |
ASTERISK-27647: app_confbridge/bridge_softmix: When channel muted report talking stopped if was talking. |
ASTERISK-27648: Timeout option for TestServer |
ASTERISK-27649: asterisk trunks showing wrong caller id |
ASTERISK-27650: Errors Using Webrtc |
ASTERISK-27651: app_confbridge: Add Muted to ConfbridgeJoin and channel snapshot headers to ConfbridgeList AMI events |
ASTERISK-27652: Null pointer Crash in PJSIP MWI |
ASTERISK-27653: asterisk 13.19.0 with the bundled pjsip installation issues |
ASTERISK-27654: Crash in ast_sip_failover_request in PJSIP |
ASTERISK-27655: Asterisk restarts unexpectedly |
ASTERISK-27656: CDR: Leaking channel snapshots allocated by stasis_channel.c |
ASTERISK-27657: res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not available) |
ASTERISK-27658: WebSocket frames with 0 sized payload causes DoS |
ASTERISK-27659: Output from rawman truncated if output is long enough |
ASTERISK-27660: Segfault in pk_atomic_get while calling pjsip_transport_destroy |
ASTERISK-27661: Add new AMI Event for Load, Unload |
ASTERISK-27662: res_pjsip: Restarting of transport does not wait long enough in all cases |
ASTERISK-27663: Register doesn’t work properly in Asterisk Realtime - not ringing calls |
ASTERISK-27664: Register doesn’t work properly in Asterisk Realtime - not ringing calls ocasionally |
ASTERISK-27665: [patch] BuildSystem: Allow fetch of PJProject without trust anchors. |
ASTERISK-27666: chan_sip: Crash processing CANCEL request |
ASTERISK-27667: Commit 8082 - 'Prune subs with reliable transports at startup' causes sourcery/contact to fill up |
ASTERISK-27668: PjSIP missing feature: Endpoint registration status |
ASTERISK-27669: [patch] codecs: Add support for WebRTC iLBC 2.0. |
ASTERISK-27670: [patch] BuildSystem: Remove chan_h323 leftovers. |
ASTERISK-27671: Deprecate legacy modules |
ASTERISK-27672: MixMonitor Audiohook SIGSEGV under Load |
ASTERISK-27673: Attended SIP Transfer via func local_attended_transfer does not call TRANSFER_CONTEXT |
ASTERISK-27674: chan_sip: RTP framing issues on outgoing calls |
ASTERISK-27675: Feature Request: Single Codec in Response |
ASTERISK-27676: Attended Transfer Music On Hold |
ASTERISK-27677: [patch] BuildSystem: Enable system provided libedit on OpenBSD. |
ASTERISK-27678: Queue Members |
ASTERISK-27679: res_pjsip: Endpoint destruction does not free DTLS configuration |
ASTERISK-27680: [patch] res_calendar: Specialized calendars depend on symbols of general calendar. |
ASTERISK-27681: [patch] BuildSystem: Enable IMAP storage on OpenBSD. |
ASTERISK-27682: PJSIP outbound-publish, ETag and Event: dialog |
ASTERISK-27683: [patch] BuildSystem: Allow newer autotools on OpenBSD. |
ASTERISK-27684: [patch] install_prereq: Update OpenBSD libraries. |
ASTERISK-27685: res_pjsip_dialog_info_body_generator: Dialog id is reused after transitioning to terminated |
ASTERISK-27686: [patch] install_prereq: Update FreeBSD libraries. |
ASTERISK-27687: FreeBSD: powl has lower than advertised precision. |
ASTERISK-27688: res_pjsip: Crash on TCP PJSIP Transport Disconnect |
ASTERISK-27689: [patch] rtp_engine: Load format name / mime type in uppercase again. |
ASTERISK-27690: SIP Session-Progress can not handled in ARI-Application |
ASTERISK-27691: Endpoint active channel count does not decrease after PickupChan |
ASTERISK-27692: bridging: Sometimes cloning the stream topology causes a crash |
ASTERISK-27693: Asterisk not sending Voicemails as email notification to users email id |
ASTERISK-27694: Asterisk not sending Voicemails as email notification to users email id |
ASTERISK-27695: res_pjsip: local_net setting not updated on config reload |
ASTERISK-27696: res_pjsip: transport option on endpoint is not enforced on incoming traffic |
ASTERISK-27697: Enable in-dialog NOTIFY on chan_pjsip channels |
ASTERISK-27698: How to install apple VOIP certificate in SIP server. |
ASTERISK-27699: How to install apple VOIP certificate in SIP server. |
ASTERISK-27700: Voicemail hangs up call on non-existent mailbox |
ASTERISK-27701: How can we create a recording folder with full permission (777) |
ASTERISK-27702: allow video/h264 breaks symmetric rtp |
ASTERISK-27703: AMI Action VoicemailUsersList returns 0 MessageCount |
ASTERISK-27704: Add cache_pools debug option to pjproject.conf |
ASTERISK-27705: chan_iax2: Stops listening for traffic |
ASTERISK-27706: PJSIP: Deadlock shutting down subscription TCP connection and sending subscription message. |
ASTERISK-27707: Segfault after hanging up a queue call where Bridge() was used |
ASTERISK-27708: I have installed the asterisk server now 10.13 64 bit |
ASTERISK-27709: [patch] BuildSystem: Avoid == for comparison in ./configure. |
ASTERISK-27710: [patch] BuildSystem: Install init scripts on openSUSE Tumbleweed. |
ASTERISK-27711: [patch] BuildSystem: Avoid re-defining of pthread_* on NetBSD. |
ASTERISK-27712: [patch] BuildSystem: Detect whether uselocale(.) is available. |
ASTERISK-27713: [patch] BuildSystem: Cast any intptr_t explicitly to its proposed type. |
ASTERISK-27714: [patch] chan_unistim: NetBSD has an incompatible struct in_pktinfo. |
ASTERISK-27715: [patch] BuildSystem: AC_PATH_PROG sets to colon character when not found. |
ASTERISK-27716: [patch] BuildSystem: Enable autotools in NetBSD. |
ASTERISK-27717: [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. |
ASTERISK-27718: [patch] BuildSystem: Enable Lua in NetBSD. |
ASTERISK-27719: [patch] res_http_post: Enable GMime in NetBSD. |
ASTERISK-27720: [patch] BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD. |
ASTERISK-27721: [patch] BuildSystem: Enable PortAudio in NetBSD. |
ASTERISK-27722: [patch] BuildSystem: Depend not implicitly but explicitly on external libraries. |
ASTERISK-27723: [patch] BuildSystem: Enable OSS Audio Emulation Library in NetBSD. |
ASTERISK-27724: I have installed and created hello world but not getting audio |
ASTERISK-27725: Avaya Deskphone display shows Calling Party Number instead of Called Party Number on Outbound Calls. |
ASTERISK-27726: chan_mobile: presents incorrect inbound Caller-ID names |
ASTERISK-27727: Asterisk not sending BYE packet to correct socket |
ASTERISK-27728: [patch] BuildSystem: Add NetBSD. |
ASTERISK-27729: [patch] install_prereq: Add NetBSD. |
ASTERISK-27730: PJSIP: Update bundled PJPROJECT to version 2.7.2 |
ASTERISK-27731: How to add a MGCP extension via FreePBX |
ASTERISK-27732: [patch] BuildSystem: Add the tool ftp to alternatively download external parts. |
ASTERISK-27733: [patch] res_srtp: Add support for libsrtp2.x on openSUSE. |
ASTERISK-27734: [patch] BuildSystem: Enable IMAP storage on openSUSE and Arch Linux. |
ASTERISK-27735: iostream.c:507 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1) |
ASTERISK-27736: [patch] install_prereq: Add SUSE. |
ASTERISK-27737: wiki: 'identify_by' pjsip.conf option documentation is wrong |
ASTERISK-27738: [patch] install_prereq: Add Arch Linux. |
ASTERISK-27739: Asterisk 13.20.0.rc2 Libresample, Hoard and MySQL |
ASTERISK-27740: chan_sip: New Channel creation from new SIP dialog with Replaces failed to be properly tracked and destroyed |
ASTERISK-27741: res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't account for more than one 'Reason' header |
ASTERISK-27742: res_monitor: Allow Monitor to Record Stereo |
ASTERISK-27743: Generic PLC doesn't work if the 2 codecs on a channel are equal |
ASTERISK-27744: Incoming queued calls are occasionally forwarded with wrong From: header |
ASTERISK-27745: [patch] BuildSystem: Remove unused dependency on libltdl. |
ASTERISK-27746: The Asterisk database manager doesn't create the field calldate under the cdr table. |
ASTERISK-27747: codec opus performance issue |
ASTERISK-27748: Random webrtc calls without audio and ALAW<->OPUS transcoding errors |
ASTERISK-27749: DIALSTATUS inconsistent |
ASTERISK-27750: app_queue.so: crash when hanging up |
ASTERISK-27751: PROJECT FOR ROIP AND TICKET MANAGEMENT |
ASTERISK-27752: Ten seconds of silence after mp3 playback |
ASTERISK-27753: bridge: Crash when freeing after bridge data after transfer occurs |
ASTERISK-27754: res_fax_spandsp: Asterisk getting "stuck" on T.38 NSF from specific vendor gateway |
ASTERISK-27755: ConfBridge: raise ConfbridgeTalking when put on hold and clear talking status |
ASTERISK-27756: bridge: Failure to impart a channel results in bad data causing crash |
ASTERISK-27757: Asterisk falling down |
ASTERISK-27758: res_rtp_asterisk: Add support for raising RTCP feedback messages |
ASTERISK-27759: res_pjsip_pubsub: Subscription persistence does not preserve XML <dialog-info> version number |
ASTERISK-27760: Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version. |
ASTERISK-27761: [patch] BuildSystem: With external editline, do not require libs for internal editline. |
ASTERISK-27763: res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream |
ASTERISK-27764: res_pjsip_pubsub: possible race condition when refreshing a subscription |
ASTERISK-27765: res_pjsip_pubsub/exten_state: two state change result in one presence notification |
ASTERISK-27766: The voice recording problem of the call that comes from queue during transmitting |
ASTERISK-27767: Asterisk got creshed |
ASTERISK-27768: [patch] app_voicemail: Two variables may be uninitialized. |
ASTERISK-27769: [patch] install_prereq: Add Gentoo Linux. |
ASTERISK-27770: [patch] install_prereq: Add Slackware (somehow). |
ASTERISK-27771: [patch] BuildSystem: Incorrect warning about /usr/lib64. |
ASTERISK-27772: i want to make the bulk calling from asterisk at a time but not able to originate the Bulk calls at a time.I want to originate the 300 to 500 calls at a time from asterisk 13 ,how can i solve this issue by using asterisk or PHP AMI |
ASTERISK-27773: Command line not being parsed correctly with getopt not from glibc |
ASTERISK-27774: res_musiconhold: Music on hold restarts after every announcement |
ASTERISK-27775: res_pjsip_notify: Multiple Event headers can be present instead of just one |
ASTERISK-27776: res_rtp_asterisk: Add support for sending RTCP feedback messages |
ASTERISK-27779: Asterisk Issue on International calls |
ASTERISK-27780: improve codec_opus |
ASTERISK-27781: Problema na instalação (app_Realtime.so) |
ASTERISK-27782: cdr_mysql: Missing MYSQL_PORT definition |
ASTERISK-27783: res_pjsip_pubsub: apparent crash on shutdown |
ASTERISK-27784: Allow for PJSIP to support using a registered IP address to do inbound IP based matching |
ASTERISK-27785: Hangup after generate call via manager with EarlyMedia |
ASTERISK-27786: app_confbridge: Add ability to enable and configure REMB support |
ASTERISK-27787: testsuite: Investigate removal of pjsua python module from all tests. |
ASTERISK-27788: pjsip does not write log file entries when an endpoint becomes (un)reachable/lagged |
ASTERISK-27789: how to add/use * and # key instead of number's 0 to 9 |
ASTERISK-27790: Realtime Queue Ringinuse=no not functioning |
ASTERISK-27791: Getting Exception "Exceptionally long voice queue length queuing to" |
ASTERISK-27792: res_pjsip: Segfault when calling pjsip_hdr_print_on in sip_msg.c:581 |
ASTERISK-27793: cppcheck identifies redundant "if" |
ASTERISK-27794: CDR: FRACK hanging up a channel transferred into ConfBridge |
ASTERISK-27795: chan_sip: one way / no audio with srtp |
ASTERISK-27796: res_hep: Allow create_address to resolve a provided hostname |
ASTERISK-27797: channel.c: BUG! Must supply a channel name or partial name to match! |
ASTERISK-27798: CDR_PROP(party_a) does not work. |
ASTERISK-27799: Voicemail messages aren't deleted when subsequently changing greeting |
ASTERISK-27800: One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP |
ASTERISK-27801: Asterisk got stuck while enabling "ari set debug all on" |
ASTERISK-27802: bad sound quality with Playback following Record |
ASTERISK-27803: MixMonitor + OGG + option "a" |
ASTERISK-27804: bridge_softmix / app_confbridge: Add support for combining REMB reports |
ASTERISK-27805: PauseMonitor Disconnects Call when using MixMonitor in Pre-Dial Handlers |
ASTERISK-27806: BASIC-RETRANS: Implement send |
ASTERISK-27807: iostreams: Potential DoS when client connection closed prematurely |
ASTERISK-27808: [patch] chan_vpb: Avoid GNU old-style field designator extension. |
ASTERISK-27809: [patch] utils/pval: Add -lBlocksRuntime for compiler clang conditionally. |
ASTERISK-27810: BASIC-RETRANS: Implement receive |
ASTERISK-27811: [patch] sip_to_pjsip: Enable python3 compatibility. |
ASTERISK-27812: When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk. |
ASTERISK-27813: REBUILD_PARSERS does not work consistently, causes build failures |
ASTERISK-27814: translate: interpolated frames are not passed through |
ASTERISK-27815: REST API - channel_ids are getting stuck |
ASTERISK-27816: func_talkdetect's logic is completely broken |
ASTERISK-27817: Infrastructure: Install python3 to all jenkins agents |
ASTERISK-27818: Username bruteforce is possible when using ACL with PJSIP |
ASTERISK-27819: Asterisk Crash - astobj2.c: FRACK!, Failed assertion bad magic number 0x0 for object 0x7fc9f402d088 |
ASTERISK-27820: [patch] Add DragonFly BSD. |
ASTERISK-27821: SEGV in res_pjsip_mwi/stasis |
ASTERISK-27822: cmd "meetme list" don't count members with Callerid Number >12 chars |
ASTERISK-27823: attended transfer from queue to queue, creates incorrect queue log entries. |
ASTERISK-27824: Fix issues exposed by GCC 8 |
ASTERISK-27825: Sometimes when the call is intercepted, the asterisk reloads |
ASTERISK-27826: res_rtp_asterisk: DTLS negotiation fails when it should succeed, causing SRTP failure |
ASTERISK-27827: Asterisk crash with match_header |
ASTERISK-27828: Add Reason to QueueCallerAbandon |
ASTERISK-27829: anonymous user login through AMI |
ASTERISK-27830: Asterisk crashes on Invalid UTF-8 string |
ASTERISK-27831: res_rtp_asterisk: Add support for abs-send-time RTP extension |
ASTERISK-27832: Asterisk Segmentation faults |
ASTERISK-27833: hangup handlers sometimes called on answer and not on hangup (depending on usage of local channels, bridge, originate, mixmonitor) |
ASTERISK-27834: PJSIP - Implement SIP_CODEC_INBOUND |
ASTERISK-27835: Asterisk Memory consumption after load |
ASTERISK-27836: res_xmpp: Disconnect does not reinstate BLF notifications |
ASTERISK-27837: GROUP_COUNT not working as expected |
ASTERISK-27838: res_rtp_asterisk.so fails to load because of a symbol defined in PJSIP, but I am using SIP |
ASTERISK-27839: Asterisk crashes due to segfault on new incoming SIP call |
ASTERISK-27840: Deadlock bridge_channel.c line 2660 (bridge_channel_internal_join) |
ASTERISK-27841: digest over for manager (ami) over http fails on too long uris |
ASTERISK-27842: Menuselect allows sanitizers to be selected even if the support libraries aren't installed |
ASTERISK-27843: res_pjsip_session.c: Wrong RTP port used on 200 OK when 180 Session Progress specifies a different port |
ASTERISK-27844: Any changes in sip.conf and then sip reload causes all call drops in case of registering sip as a client to tollfree providers |
ASTERISK-27845: Codec-Change Re-INVITE during DTMF can cause marker bit error |
ASTERISK-27846: ast_coredumper: Fix OUTPUT directory |
ASTERISK-27847: Asterisk Crashes (Excessive refcount) |
ASTERISK-27848: rtp: DTMF Breaks With telephony-event/16000 |
ASTERISK-27849: Transport=udp parameter is not present in SIP INVITE request URI |
ASTERISK-27850: [patch] rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. |
ASTERISK-27851: app_confbridge: Opus participants have bad quality in confbridge audio conference with non-20ms mixing interval |
ASTERISK-27852: cli: "manager show settings" mislabels HTTP timeout as being minutes. |
ASTERISK-27853: Incorrect error reported when leaving/retrieving a ODBC voicemail |
ASTERISK-27854: rtp: Crash in off-nominal case where RTP instance can't be set up |
ASTERISK-27855: Dial from macro-context N-way conference |
ASTERISK-27856: Dial from macro-context N-way |
ASTERISK-27857: Attended Transfer: Attended transfer has failed if using AMI terminal to send. |
ASTERISK-27858: Cannot build static with chan_mgcp, chan_phone |
ASTERISK-27859: Crash when freeing after bridge |
ASTERISK-27860: [patch] res_pjsip: Register pjsip_transport_management not externally but internally. |
ASTERISK-27861: [patch] res_pjsip_endpoint_identifier_ip: Unregister the module for headers. |
ASTERISK-27862: ARI: Crashing on json_mem_free (after sending event) |
ASTERISK-27863: config/ast_destroy_realtime_fields: successful DELETE is treated as failed |
ASTERISK-27864: Create NOTICE for INVITES with no matching peer |
ASTERISK-27865: [patch]: tcptls: Repair ./configure --with-ssl=PATH. |
ASTERISK-27867: [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated. |
ASTERISK-27868: cleanup: several null pointer dereference (detected by cppcheck) |
ASTERISK-27869: couple of memory leaks (detected by cppcheck) |
ASTERISK-27870: app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts |
ASTERISK-27871: Remote URL in playback must end with file extension |
ASTERISK-27872: res_pjsip: Modified qualify_frequency doesn't effect until pjsip reload |
ASTERISK-27873: documentation: Error on wiki description of Asterisk 13 "MeetmeMute" event |
ASTERISK-27874: [patch] tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. |
ASTERISK-27875: AMI Hook Helper Problem |
ASTERISK-27876: [patch] tcptls: Allow OpenSSL configured with no-dh. |
ASTERISK-27877: app_confbridge: Add talking indicator for ConfBridgeList AMI response |
ASTERISK-27878: [patch] tcptls.h: Repair ./configure --with-ssl=PATH. |
ASTERISK-27879: pjsip / SRV uses wrong IP address for OPTIONS packages / qualify |
ASTERISK-27880: [patch] pjproject_bundled: Repair ./configure --with-ssl=PATH. |
ASTERISK-27881: PBX calls via chan_sip TCP trunk now get authentification error |
ASTERISK-27882: Do Asterisk support emergency call for more than 20 seconds? |
ASTERISK-27883: No voice while using sipml5 in IPv6 environment |
ASTERISK-27884: asterisk early processing the completion of the transferred call |
ASTERISK-27885: Calls to SIP phones fail to establish, debug log and tcpdump suggest a packet backlog |
ASTERISK-27886: Crash Asterisk 13.21.0 during SRTP |
ASTERISK-27887: func_channel: Accessing CHANNEL(pjsip,call-id) may crash if the channel hangs up. |
ASTERISK-27888: SQL fetch error on query which return 0 columns |
ASTERISK-27889: Error to start dahdi |
ASTERISK-27890: Sip to PJSIP conversion scripts errors |
ASTERISK-27891: Sip to PJSIP conversion scripts errors |
ASTERISK-27892: No audio with IPV6 enabled ENDPOINT |
ASTERISK-27893: testsuite: Current version of the Python module construct is incompatible. |
ASTERISK-27894: [patch] testsuite: Avoid KeyError when astdatadir not set in asterisk.conf. |
ASTERISK-27895: chan_pjsip: 'tel' URI is unsupported |
ASTERISK-27896: tests/channels/SIP/SDP_attribute_passthrough: Requires codec_speex. |
ASTERISK-27897: tests/manager/acl-login: An AMI Login was allowed which was not expected. |
ASTERISK-27898: Crash at the and of a call |
ASTERISK-27899: res_pjsip: XMPP Status to OpenFire incorrect after OPTIONs rewrite |
ASTERISK-27900: [patch] tests/codecs/opus/{de,en}code: Requires python-numpy. |
ASTERISK-27901: [patch] ooh323c: GCC 8: output truncated before terminating nul. |
ASTERISK-27902: chan_pjsip isn't updating hangupcause on 4XX responses |
ASTERISK-27903: menuselect: GCC 8: restrict-qualified parameter passed and aliased. |
ASTERISK-27904: [patch] testsuite: Enable asttest in (Red Hat) Fedora. |
ASTERISK-27905: [patch] res_srtp: Repair ./configure --with-ssl=PATH. |
ASTERISK-27906: [patch] res_crypto: Allow OpenSSL configured with no-deprecated. |
ASTERISK-27907: Active calls dropped randomly on Asterisk |
ASTERISK-27908: [patch] crypto.h: Repair ./configure --with-ssl=PATH. |
ASTERISK-27909: cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch |
ASTERISK-27910: [patch] res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. |
ASTERISK-27911: Deadlock, likely when delegating calls from queue with Redirect |
ASTERISK-27912: [PATCH] Add predial handler to app_queue |
ASTERISK-27913: chan_sip / chan_pjsip: T.38 fails if SDP negotiation results in declined stream |
ASTERISK-27914: [patch] tests/test_utils: Repair ./configure --with-ssl=PATH. |
ASTERISK-27915: menuselect: List of dependencies cut too early (~35 chars). |
ASTERISK-27916: Error 4 in app_queue.so in 13.18-cert3 |
ASTERISK-27917: Asterisk crashes when second channel is added to bridge (using pjsip) |
ASTERISK-27918: Transfer in U&D mode |
ASTERISK-27920: app_queue: Queue member considered inuse after immediately hanging up during dialing. |
ASTERISK-27921: Asterisk crash |
ASTERISK-27922: Force AST_FRAME_DTMF_END on AST_FRAME_DTMF_BEGIN |
ASTERISK-27923: Crash when realtime resouce_list is enabled |
ASTERISK-27924: $agi->get_data( not play File and no Response |
ASTERISK-27925: [patch] tests/channels/SIP/rfc2833_dtmf_detect: Requires rawsocket. |
ASTERISK-27926: [patch] bootstrap.sh: find -maxdepth is not POSIX compatible. |
ASTERISK-27927: res_xmpp: <c ver='asterisk-xmpp'> not XEP-115 conformant |
ASTERISK-27928: segfault in channel_read_pjsip, dereferencing chan |
ASTERISK-27929: [patch] BuildSystem: Enable autotools in Solaris 11. |
ASTERISK-27930: res_pjsip: PJSIP TCP Segfault. |
ASTERISK-27931: [patch] BuildSystem: Enable ./configure in Solaris 11. |
ASTERISK-27932: [patch] tests/yappcap: Requires rawsocket. |
ASTERISK-27933: [patch] uuid: Enable UUID in Solaris 11. |
ASTERISK-27934: Program terminated with signal 11, Segmentation fault. |
ASTERISK-27935: app_voicemail: emailbody per user can't contain commas |
ASTERISK-27936: res_pjsip_session doesn't update media when a 200 comes in with a different port than a 183 |
ASTERISK-27937: build: POSIX does not know sed -i |
ASTERISK-27938: [patch] Compile fails with `IPTOS_MINCOST' undeclared. |
ASTERISK-27939: [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11. |
ASTERISK-27940: asterisk crash randomly |
ASTERISK-27941: more than required parameters in asterisk application Record is working weirdly |
ASTERISK-27942: res_pjsip_messaging doesn't accept application/* content-types. |
ASTERISK-27943: AMI: Action SendText needs to use the correct thread. |
ASTERISK-27944: res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE |
ASTERISK-27945: TCP-Peer with insecure=port not found on incoming call |
ASTERISK-27946: dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't |
ASTERISK-27947: Asterisk and Siemens HiPath8000 Integration |
ASTERISK-27948: chan_phone fails to build: ixjuser.h removed in kernel >=4.17 |
ASTERISK-27949: res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers |
ASTERISK-27950: Context Visability between AEL and Extensions |
ASTERISK-27951: Segfault in res_pjsip_t38.so on incoming fax |
ASTERISK-27952: Segfault after pjsip hdr linked list corruption |
ASTERISK-27953: two consecutive spaces |
ASTERISK-27954: Queue Log "CONNECT" event incorrect "ringtime" value |
ASTERISK-27955: res_pjsip_session: sdp group:BUNDLE attribute truncated |
ASTERISK-27956: res_pjsip_pubsub: segfault in function publish_expire |
ASTERISK-27957: PJSIP proposes ICE candidates on answer even if not in offer |
ASTERISK-27958: Deadlock in res_pjsip_registrar on contact transport shutdown |
ASTERISK-27959: [patch] Asterisk 15.4.1 h264 fmtp negotiation problem |
ASTERISK-27960: Testing workflow change |
ASTERISK-27961: res_pjsip: Spurious ERROR logging when printing headers in sip_msg |
ASTERISK-27962: res_pjsip_t38: Reinvite after 491 includes audio stream and T.38, not just T.38 |
ASTERISK-27963: res_pjsip: ignore Path header on INVITE |
ASTERISK-27964: app_queue: ring_entry accesses nativeformats without channel lock or reference |
ASTERISK-27965: module: Remove old modules, update support levels |
ASTERISK-27966: pjsip: Race condition in 183 re transmission can result in a deadlock |
ASTERISK-27967: srtp: rejecting short sdes lifetimes incompatible with obihai ATAs |
ASTERISK-27968: systemd: asterisk.service |
ASTERISK-27970: res_rtp_asterisk: T.140 packets containing backspace or end of line are merged with regular text and it causes some UA to break |
ASTERISK-27971: res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability |
ASTERISK-27972: res_sorcery_config: Allow object name based matching |
ASTERISK-27973: app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY |
ASTERISK-27974: core: Leaksanitizer reports libedit2 el_set EL_ADDFN wcsdup leaks |
ASTERISK-27975: Command order to pull the second line |
ASTERISK-27976: Command order to pull the second line |
ASTERISK-27977: Crash at T38 call disconnection |
ASTERISK-27978: res_pjsip: Change default transport keepalive to preserve behavior |
ASTERISK-27979: externals: Script assumes that architecture is Intel |
ASTERISK-27980: Caller ID cannot be changed on Attended Transfer before dialing out |
ASTERISK-27981: res_fax: Fax session leak with fax gatewaying |
ASTERISK-27982: res_pjsip: Crash on improperly formatted match_header in ps_endpoint_ips |
ASTERISK-27983: pjsip_options: rework may have left concurrency issue |
ASTERISK-27984: PJSIP: Stuck channel when using conference on an LTE connection -> Wifi |
ASTERISK-27985: PJSIP: Does not respond to INVITES when any taskprocessor queue length is > high water level |
ASTERISK-27986: app_speech_utils: Assertion when creating |
ASTERISK-27987: AGI() call causes RTP to briefly freeze under certain circumstances |
ASTERISK-27988: alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean |
ASTERISK-27989: res_pjsip_sdp_rtp: Asterisk sources RTP from wrong IPv6 address |
ASTERISK-27990: res_rtp_asterisk: Requires OpenSSL in Developer Mode. |
ASTERISK-27991: BuildSystem: Enable Jansson in Solaris 11. |
ASTERISK-27992: PJSIP: Adding `sends_registrations = yes` to pjsip_wizard.conf causes crash |
ASTERISK-27993: pjsip_wizard example gives wrong info about unsupported SRV records |
ASTERISK-27994: PJSIP: Early media ringback not indicated after Progress() |
ASTERISK-27995: pjproject_bundled: Find shared libraries in root --with-ssl=PATH. |
ASTERISK-27996: PJSIP: match_header endpoint matching does not support inbound registration |
ASTERISK-27997: pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. |
ASTERISK-27998: Segfault after REFER packet |
ASTERISK-27999: Wrong SRTP use status report |